search for: update_call_count

...retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes monitor-format = wav ringinuse = no I am using AddQueueMember to add SIP interface to the queue. Each sip interface is member of multiple queues. Occasionally I get messages like [Mar 17 11:33:01] ERROR[9253]: chan_sip.c:3232 update_call_counter: Call to peer '2505' rejected due to usage limit of 1 [Mar 17 11:33:01] ERROR[9254]: chan_sip.c:3232 update_call_counter: Call to peer '2509' rejected due to usage limit of 1 [Mar 17 11:33:01] ERROR[9255]: chan_sip.c:3232 update_call_counter: Call to peer '2502' rejected d...

...ot;Unknown" to be a valid state to dispatch a caller to. So my agents start getting flooded with calls while already on the phone, then the call-limit I've configured in sip.conf kicks in and my console fills up with this: pbxtel-01*CLI> [Aug 29 16:44:04] ERROR[22621]: chan_sip.c:3169 update_call_counter: Call to peer '1405' rejected due to usage limit of 2 pbxtel-01*CLI> [Aug 29 16:44:04] ERROR[22621]: chan_sip.c:3169 update_call_counter: Call to peer '1410' rejected due to usage limit of 2 pbxtel-01*CLI> [Aug 29 16:44:04] ERROR[22762]: chan_sip.c:3169 update_call_counter...

...ur pbx but since no voice was passing I decided to go back to old version (0.3.1-rc23). Last friday everything seemed to work fine but now every incoming call drops after 3-4 seconds while Asterisk console is showing these messages: Apr 23 12:42:39 DEBUG[7625]: chan_sip.c:2426 sip_hangup: update_call_counter(3) - decrement call limit counter Apr 23 12:42:39 DEBUG[7625]: chan_sip.c:1379 __sip_ack: Acked pending invite 102 Apr 23 12:42:39 DEBUG[7625]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '1fd7824840123666030e29a70d1d7739@192.168.1.200' of Request 102: Match Found...

Guys. Ive seen on my asterisk messages log that asterisk has shutdown itself about 12 times in 5 days... The logs show nothing but: [Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Drop call [Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Release call [Jun 28 09:40:02] VERBOSE[3172]: [Jun 28 09:40:02] -- Unicall/4 released [Jun 28 09:40:02] VERBOSE[3084]: [Jun 28 09:40:02] Asterisk cleanly

...s SIP/232-2e41 and SIP/cerved-out-6eba Jun 14 16:25:14 DEBUG[14380] channel.c: Hanging up channel 'SIP/cerved-out-6eba' Jun 14 16:25:14 DEBUG[14380] chan_sip.c: Hangup call SIP/cerved-out-6eba, SIP callid 362258b02bbafa8117eecbb6755837a0@10.97.1.254) Jun 14 16:25:14 DEBUG[14380] chan_sip.c: update_call_counter(9704) - decrement call limit counter Jun 14 16:25:14 DEBUG[14380] chan_sip.c: Updating call counter for outgoing call Jun 14 16:25:14 DEBUG[14380] app_dial.c: Exiting with DIALSTATUS=ANSWER. I have Asterisk 1.2.8 but remote server has 1.2.4. Any help? -- Domenico Viggiani

Hi, Anyone can tell me what this means? [Feb 3 12:42:32] WARNING[12130]: chan_sip.c:3293 update_call_counter: Inringing for peer 'test-peer' < 0? Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090203/a39503b7/attachment.htm

...7 at default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic busylevel=1 limitonpeers=yes call-limit=1 when 7527 is busy i am getting following error message on CLI. Why i am getting channel status CONGESTION ? instead BUSY ? [May 2 16:47:31] NOTICE[31757]: chan_sip.c:5658 update_call_counter: Call to peer '7527' rejected due to usage limit of 1 -- Couldn't call 7527 -- Called 7527 == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s at macro-stdexten:2] Goto("SIP/7604-00000006", "s-CONGESTION,1") in new stack -- Goto (macr...

...n call-limit reached dialstatus is CHANUNAVAIL and CDR(disposition)='NO ANSWER' -- Executing [0014 at sub_pbxdialco:49] Dial("SIP/1295-000001f8", "SIP/0014,12,tTkK") in new stack == Using SIP RTP CoS mark 5 [2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003 update_call_counter: Call to peer '0014' rejected due to usage limit of 1 -- Couldn't call 0014 == Everyone is busy/congested at this time (0:0/0/0) -- Executing [0014 at sub_pbxdialco:50] NoOp("SIP/1295-000001f8", "CHANUNAVAIL") in new stack I think that isn't co...

...nversation, the peer call counter is set to 2.The problem is that, the counter is not reset to zero after hangup and becoz of this the user is not able to recieve any call anymore even if s/he has hungup. the asterisk cli displays the following error. [Mar 17 16:15:10] ERROR[7664]: chan_sip.c:3030 update_call_counter: Call to peer 'rehmat' rejected due to usage limit of 2 -- Couldn't call rehmat == Everyone is busy/congested at this time (0:0/0/0) Im using asterisk1.4.0 . declaring type=peer solves the problem. but if anybody knows why its not working for type=friend, plz share. -- Regard...

Hi I have noticed that asterisk returns 'SIP 603' when the called party does not answer. My test setup is simple: two SIP phones (extensions: 100 and 111) registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds. When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to 111 (expected) and a '603 Decline' response to 100 (unexpected &

Hello All I am trying to answer incoming SIP calls, first, by dialing an extension, thence into voicemail, which works; and secondly by going straight into voice mail which does not. The extension.conf that works is like this; [incomingSIP] exten=>_.,1,Dial,Zap/2|1 exten=>_.,2,Voicemail,u5152 exten=>_.,3,Hangup the extension.conf which does not is like this; [incomingSIP]

...IP/51-0986fab0 Jan 31 15:20:47 DEBUG[2442] channel.c: Done planning to masquerade channel SIP/53-b7a05818 into the structure of SIP/51-0986fab0 Jan 31 15:20:47 DEBUG[25962] channel.c: Got clone lock for masquerade on 'SIP/53-b7a05818' at 0xb7a0adf4 Jan 31 15:20:47 DEBUG[25962] chan_sip.c: update_call_counter(51) - decrement call limit counter Jan 31 15:20:47 DEBUG[25962] chan_sip.c: Acked pending invite 102 Jan 31 15:20:47 DEBUG[25962] chan_sip.c: Stopping retransmission on '340302fd7e29e92250436f617fcfdd03@10.0.0.60' of Request 102: Match Found Jan 31 15:20:47 DEBUG[25962] channel.c: Putti...

...VERBOSE[29697] logger.c: -- Called 7190 Jun 16 15:48:05 VERBOSE[29691] logger.c: -- Agent/7132 is ringing Jun 16 15:48:20 VERBOSE[29697] logger.c: -- Nobody picked up in 15000 ms Jun 16 15:48:20 DEBUG[29691] app_queue.c: Dunno what to do with control type -1 Jun 16 15:48:20 DEBUG[29697] chan_sip.c: update_call_counter(7190) - decrement call limit counter Jun 16 15:48:20 DEBUG[29697] chan_sip.c: Acked pending invite 102 un 16 15:48:20 DEBUG[29697] app_dial.c: Exiting with DIALSTATUS=NOANSWER. Jun 16 15:48:20 DEBUG[29697] pbx.c: Expression result is '1' Jun 16 15:48:20 VERBOSE[29691] logger.c: -- Agent/7...

...ject for frame 0, retransmitting frame 0 now, updating n_r! !! Got reject for frame 0, retransmitting frame 1 now, updating n_r! Echo cancellation already on == Primary D-Channel on span 1 down No D-channels available! Using Primary channel 24 as D-channel anyway! == Primary D-Channel on span 1 up update_call_counter(1630) - decrement call limit counter

...: Found Feb 8 21:20:18 DEBUG[4160] channel.c: Avoiding initial deadlock for 'SIP/1234-3a53' Feb 8 21:20:18 VERBOSE[6600] logger.c: -- SIP/1234-3a53 is ringing Feb 8 21:20:19 VERBOSE[4179] logger.c: -- Channel 0/12, span 4 got hangup request Feb 8 21:20:19 DEBUG[6600] chan_sip.c: update_call_counter(1234) - decrement call limit counter Feb 8 21:20:19 DEBUG[6600] chan_sip.c: Acked pending invite 102 Feb 8 21:20:19 DEBUG[6600] chan_sip.c: Stopping retransmission on '1187ae7d6be6e0761d99cf5245a9058d@202.58.255.137' of Request 102: Match Found Feb 8 21:20:19 DEBUG[6600] chan_sip.c: St...

Hi, My asterisk sometimes stop responding to iax calls. In the log, I've found this: Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[13002] chan_sip.c: update_call_counter(1409) - decrement call limit counter Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadl...

...m 192.168.2.13 Mar 26 09:51:18 DEBUG[3896] chan_sip.c: Hairpin detected, setting up call forward for what it's worth Mar 26 09:51:18 VERBOSE[4885] logger.c: -- Now forwarding Zap/55-1 to 'Local/201@from-sip' (thanks to SIP/192.168.2.13-08e24bd0) Mar 26 09:51:18 DEBUG[4885] chan_sip.c: update_call_counter(201) - decrement call limit counter After that it will loop hundreds of times with a block like this in the log: Mar 26 09:51:18 VERBOSE[4888] logger.c: -- Executing Goto("Local/201@from-sip-661a,2", "to-sip|201|1") in new stack Mar 26 09:51:18 VERBOSE[4888] logger.c:...

...tp.c: Ooh, format changed from unknown to ulaw Oct 16 11:01:17 DEBUG[6775] channel.c: Hanging up channel 'SIP/snom3-08984140' Oct 16 11:01:17 DEBUG[6775] chan_sip.c: Hangup call SIP/snom3-08984140, SIP callid 281fa14d186c76c5545b59d8253762ca@wx3.se) Oct 16 11:01:17 DEBUG[6775] chan_sip.c: update_call_counter(snom3) - decrement call limit counter Oct 16 11:01:17 DEBUG[6775] chan_sip.c: Updating call counter for outgoing call Oct 16 11:01:17 DEBUG[6767] rtp.c: Got RTCP report of 52 bytes Oct 16 11:01:17 DEBUG[6767] app_meetme.c: Got unrecognized frame on channel SIP/snom1-b7d0c328, f->frametype=5...

...51-0986fab0 Jan 31 15:20:47 DEBUG[2442] channel.c: Done planning to masquerade channel SIP/53-b7a05818 into the structure of SIP/51-0986fab0 Jan 31 15:20:47 DEBUG[25962] channel.c: Got clone lock for masquerade on 'SIP/53-b7a05818' at 0xb7a0adf4 Jan 31 15:20:47 DEBUG[25962] chan_sip.c: update_call_counter(51) - decrement call limit counter Jan 31 15:20:47 DEBUG[25962] chan_sip.c: Acked pending invite 102 Jan 31 15:20:47 DEBUG[25962] chan_sip.c: Stopping retransmission on '340302fd7e29e92250436f617fcfdd03@10.0.0.60' of Request 102: Match Found Jan 31 15:20:47 DEBUG[25962] channel.c: Pu...

...uting [xxxxxxxxxx at from-internal:1] Macro("SIP/6001-007e2840", "trunkdial|+1xxxxxxxxxx") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/6001-007e2840", "SIP/trunk1/+1xxxxxxxxxx") in new stack [Sep 10 09:06:52] ERROR[16253]: chan_sip.c:3192 update_call_counter: Call to peer 'trunk1' rejected due to usage limit of 1 -- Couldn't call trunk1/+1xxxxxxxxxx == Everyone is busy/congested at this time (0:0/0/0) -- Executing [s at macro-trunkdial:2] Hangup("SIP/6001-007e2840", "") in new stack I don't want the dial...